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Fix caps on audio pipeline 

Fixes #1747

The pipeline has the caps for the analyzer applied in the wrong place. This results in the audio output being limited to 16 bit regardless of the input file.
This change also cleans up the mono/sample rate caps as well.
This commit is contained in:
Mark Furneaux 2016-05-15 13:31:00 -04:00
parent e31278c056
commit bcf29dc670
1 changed files with 20 additions and 25 deletions
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45
src/engines/gstenginepipeline.cpp
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@ -393,13 +393,7 @@ bool GstEnginePipeline::Init() {
}
gst_element_link_many(queue_, audioconvert_, convert_sink, nullptr);
// Link the elements with special caps
// The scope path through the tee gets 16-bit ints.
GstCaps* caps16 = gst_caps_new_simple("audio/x-raw", "format", G_TYPE_STRING,
"S16LE", NULL);
gst_element_link_filtered(probe_converter, probe_sink, caps16);
gst_caps_unref(caps16);
gst_element_link(probe_converter, probe_sink);
// Link the outputs of tee to the queues on each path.
gst_pad_link(gst_element_get_request_pad(tee, "src_%u"),
@ -412,31 +406,32 @@ bool GstEnginePipeline::Init() {
gst_element_link_many(rgvolume_, rglimiter_, audioconvert2_, tee, nullptr);
}
// Link everything else.
gst_element_link(probe_queue, probe_converter);
// Link the analyzer output of the tee and force 16 bit caps
GstCaps* caps16 = gst_caps_new_simple("audio/x-raw", "format", G_TYPE_STRING,
"S16LE", NULL);
gst_element_link_filtered(probe_queue, probe_converter, caps16);
gst_caps_unref(caps16);
gst_element_link_many(audio_queue, equalizer_preamp_, equalizer_,
stereo_panorama_, volume_, audioscale_, convert,
nullptr);
// add caps for fixed sample rate and mono, but only if requested
if (sample_rate_ != GstEngine::kAutoSampleRate && sample_rate_ > 0) {
GstCaps* caps = gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT,
sample_rate_, nullptr);
if (mono_playback_) {
gst_caps_set_simple(caps, "channels", G_TYPE_INT, 1, nullptr);
}
// Ensure that the audio output of the tee does not autonegotiate to 16 bit
GstCaps* caps = gst_caps_new_simple("audio/x-raw", "format", G_TYPE_STRING,
"F32LE", NULL);
gst_element_link_filtered(convert, audiosink_, caps);
gst_caps_unref(caps);
} else if (mono_playback_) {
GstCaps* capsmono =
gst_caps_new_simple("audio/x-raw", "channels", G_TYPE_INT, 1, nullptr);
gst_element_link_filtered(convert, audiosink_, capsmono);
gst_caps_unref(capsmono);
} else {
gst_element_link(convert, audiosink_);
// Add caps for fixed sample rate and mono, but only if requested
if (sample_rate_ != GstEngine::kAutoSampleRate && sample_rate_ > 0) {
gst_caps_set_simple(caps, "rate", G_TYPE_INT, sample_rate_, nullptr);
}
if (mono_playback_) {
gst_caps_set_simple(caps, "channels", G_TYPE_INT, 1, nullptr);
}
gst_element_link_filtered(convert, audiosink_, caps);
gst_caps_unref(caps);
// Add probes and handlers.
gst_pad_add_probe(gst_element_get_static_pad(probe_converter, "src"),
GST_PAD_PROBE_TYPE_BUFFER, HandoffCallback, this, nullptr);